南華大學機構典藏系統:Item 987654321/26402
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    题名: 以近似class-2與class-2 ANSI S1.11 濾波器組設計為基礎之噪音抑制演算處理器研製(I)
    其它题名: An Effective Noise Reduction Algorithm and Its Chip Design Based on Quasi-Class-2 and Class-2 Ansi S1.11 Filterbanks
    作者: 賴信志
    貢獻者: 南華大學資訊工程學系
    日期: 2017
    上传时间: 2018-12-06 15:50:15 (UTC+8)
    摘要: 本研究提出以頻帶為基礎的噪音抑制演算法,分別結合離散餘弦轉換(Discrete Cosine Transform)與離散傅立葉轉換(Discrete Fourier Transform)為核心架構實現的近似 Class-2、Class-2、以及Class-0 ANSI S1.11 1/3 八度音濾波器組。核心主要之技術為:(1)改善各頻帶間的不匹配現象,亦希望能獲得較低的群延遲與較低的計算複雜度,以及能有效降低聽力圖上的匹配誤差;(2)利用交互相關性為基準的聲源定位演算法,來避免音源能量過強導致判斷上的錯誤;(3)搭配麥克風通道分別計算到的功率頻譜密度分析後進行感知決策。相關之技術研發有助於高規格助聽輔具的設計與發展。
    This research mainly focuses on developing frequency-subband based noise reduction algorithm, and it integrates with quasi-class-2, class-2, and class-0 ANSI S1.11 1/3-octave filterbank design based on the DCT and DFT modulations, respectively. The kernel techniques involves: (1) Improving the delay mismatch of each subband, and achieving shorter group delay, lower computational complexity, and lower matching error of the audiogram; (2) Using Cross-correlation to realize sound-source localization and avoiding the problem of more sound-source energy more decision error; (3) Calculating and analyzing the power spectrum density of every microphone, and then accomplishing awareness-computation strategy. This research would bring more design concepts and application on the topic of hearing aids, and the proposed method would be more suitable for high-level applications of hearing aids in the future.
    显示于类别:[資訊工程學系] 國科會計畫

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